[Libav-user] filtering_audio.c example not working

Ron Woods rwoods at vaytek.com
Mon Nov 12 21:50:24 CET 2012

I am trying out the filtering_audio.c example provided with the ffmpeg libraries for Windows to extract the audio from a MP4 file, resample to 8 KHz and convert from stereo to mono. The example pipes the audio output via stdout to ffplay. I am using Visual Studio 2010 and the example successfully builds and runs but the result is clearly not the desired result. At the end of init_filters I added a call to avfilter_graph_dump() and it all looks correct and also the pipe to ffplay as in this trace:

abuffer filter args: time_base=1/24000:sample_rate=24000:sample_fmt=s16:channel_layout=0x4
Output: srate:8000Hz fmt:s16 chlayout:mono
|    in     |default--[24000Hz s16:mono]--Parsed_aconvert_0:default
| (abuffer) |

Parsed_aresample_1:default--[8000Hz s16:mono]--default|       out       |
                                                      | (ffabuffersink) |

in:default--[24000Hz s16:mono]--default| Parsed_aconvert_0 |default--[24000Hz s16:mono]-Parsed_aresample_1:default
                                       |    (aconvert)     |

Parsed_aconvert_0:default--[24000Hz s16:mono]--default| Parsed_aresample_1 |default--[8000Hz s16:mono]--out:default
                                                      |    (aresample)     |
[s16le @ 003edda0] Invalid sample rate 0 specified using default of 44100
[s16le @ 003edda0] Estimating duration from bitrate, this may be inaccurate
Input #0, s16le, from 'pipe:':
  Duration: N/A, start: 0.000000, bitrate: 128 kb/s
    Stream #0:0: Audio: pcm_s16le, 8000 Hz, 1 channels, s16, 128 kb/s

If you have made this example run properly on Windows in VS 2010, would you please provide any tips or changes you made for it to work?

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